Introduction

A hands-free telephone or full-duplex intercom system has a feedback or echo problem

because the output from the loudspeaker feeds into the microphone

Echo is a delayed and distorted version of an original sound or electrical signal which is
reflected back to the source.

In telecommunication networks there are two type of echo.
Line Echo
Acoustic Echo

Line Echo

The electrical echo is due to the impedance mismatch.
Found in PSTN, mobile, and IP phone systems. The electric
echo is created at the hybrid connections. Fig 1: Hybrid Echo

Acoustic Echo

The acoustic echo is due to the coupling between the
loudspeaker and microphone. Found in mobiles,
videophone terminals, in teleconferencing etc.
Generation of acoustic echo through direct coupling and reverberations.
Fig 2: Acoustic echo Echo


Hands free environment

Fig1: Basic set up of a Hands Free Communication System

Near-End Signal (Speaker)

x(n): far-end signal

converted into sound by a
speaker configuration

propagates within a given room
y(n) = conv(x(n),h(n))
Near-End Signal (Receiver)
d(n): microphone signal

captures the sound within the
room

input back into the electronic
system

composed of echo, near-end
signal, and extraneous noise
Echo cancellation algorithm
Principle of AEC:
To remove the acoustic echo from the circuit, echo cancellation creates a model of
the echo path, synthesizes a replica estimate of the echo and cancels the echo by
subtracting the estimated echo from the true echo.
The modeling and subtraction process involves taking a sample of the received
signal, storing the sample in a convolution processor, and creating a model of the
echo using echo canceller,
When any reflected signals pass back through the convolution processor, the
canceller recognizes these signals as undesirable echo and subtracts the echo
model from the actual signal to cancel out the echo.
A simple way to achieve these processes is using an FIR filter. However, such an
echo cancellation scheme is naturally linear and is very sensitive to the
nonlinearities in the echo path. A better performance can be achieved by using the
adaptive algorithms.


Fig 2: echo cancellation algorithm
Acoustic Echo Canceller with Doubletalk Detector and
Nonlinear Processor
Fig 3: Echo Canceller with Doubletalk Detector and Nonlinear Processor
Adaptive filter
Adaptive filter are mainly used in cases where signal conditions or system
parameters are slowly changing and filter is adjusted to compensate for this
change.

a filter is a device that maps its input signal into another output signal by
extracting only the desired information contained in the input signal.

adaptive filters are time varying since their parameters are continually
changing in order to meet a performance requirement.

The coefficients of the adaptive filter are adjusted, or optimized, using an
LMS algorithm based on the error signal.

Adaptive filter algorithms
Lms algorithm

Adaptive algorithm in which the solution is found by minimizing the mean error between
the desired solution and the computed observed data.

It is stochastic gradient descent method in that the filter is only adapted based on the error at
correct time.


Drawback:

Lms is sensitive to scaling of its input x(n)
This makes its very hard to choose a
step size that guarantees stability of algorithm





















1.Normalized Least Mean Square (NLMS) Algorithm
This algorithm is variant of lms algorithm that solves by normalizing with the power
of the input .

The normalized LMS(NLMS), algorithm utilizes a variable convergence factor that
minimizes the instantaneous error, Such a convergence factor usually reduces the
convergence time.

The updating equation of the LMS algorithm can employ a variable convergence
factor k in order to improve the convergence rate. In this case, the updating
formula is expressed as
w(k +1) = w(k) + 2 k e(k)x(k) = w(k) + w (k),
where k must be chosen with the objective of achieving a faster convergence.

Using the variable convergence factor the updating equation for the NLMS
algorithm is given by
w(k+1) = w(k)*(e(k)*x(k) /x^T(k)*x(k)

Drawback: fast convergence and low excess MSE for NLMS with fixed regularization
parameter.
1.Normalized Least Mean Square (NLMS) Algorithm
2.Variable step size NLMS algorithm
Both the LMS and the NLMS algorithms have a fixed step size value for
every tap weight in each iteration. The VSS-NLMS adaptive algorithms
provide an efficient solution to the conflicting requirements of fast
convergence and low maladjustment needed by any adaptive system

The use of the VSS-NLMS algorithm will eliminate much of the trade-off
between residual error and speed of convergence existing with the fixed
step-size NLMS algorithm and therefore resulting in an improved
performance.

Given the input vector, the Euclidean norm of the input vector ,
the NLMS algorithm with fixed step size, , for adjusting the adaptive echo
cancellers coefficients at time instant k is defined as follows:


Variable step size NLMS algorithm










In this method, the fixed step size in (1) is made variable and is updated according
to the following recursion:

Which can be transformed into



where the parameter is a small positive constant that controls the adaptive behavior of the
step-size sequence
k and T denotes transpose operation.
Accordingly, the coefficients of the VSS-NLMS echo canceller will be updated
according to a variable step-size NLMS (VSSNLMS) algorithm given by



where the variable step size parameter is confined to the following limits:

min, max are chosen to satisfy the
convergence requirements on nlms
with fixed step size 0 < min < max < 2.

Other VS-NLMS algorithms
Generalized Normalized Gradient Descent algorithm (GNGD)
algorithm
Robust Regularization (RR) -NLMS Algorithm
The Generalized Square-Error Regularized (GSER) Algorithm






Generalized Normalized Gradient Descent algorithm (GNGD) algorithm


The GNGD belongs to the family of time-varying regularized VSS algorithm.

The filter coefficient vector is updated as


where
c
is a fixed step size, and the regularization parameter (n) is recursively
calculated as



where is an adaptation parameter needs tuning, and the initial value (0) has to be
set as well.
Drawback: (GNGD) algorithm uses a time-varying regularization parameter, i.e.,
the regularization parameter updated at every iteration. The update scheme for
the regularization parameter works well in the initial transient phase but shows
limited performance as the iteration goes. This is because the adaptation of the
regularization parameter is not robust enough to cope with the varying state of the
adaptive filter.







Robust Regularization (RR) -NLMS Algorithm
To make the regularization parameter robust, we introduce the normalized
gradient into the update scheme for the regularization parameter. proposed
method outperforms the -NLMS and the GNGD in terms of the
convergence speed and the misadjustment error for linear, nonlinear, and
speech input signals.

RR-NLMS algorithm is a modified version of GNGD. The regularization
parameter is updated as





where sgn(x) represents the sign function, and

min
is a parameter needs tuning.





The Generalized Square-Error Regularized (GSER) Algorithm
new regularized NLMS algorithm outperforms existing schemes in
convergence, tracking, and mis-adjustment.
The GSER updates w(n) as follows



where is a positive parameter that makes the filter more general, and the
power of the error signal is estimated.


Gauss-Seidel Pseudo Affine Projection algorithm (GSPAP)
stable, fast convergent and has good tracking abilities.
improved performances and more complex than the NLMS algorithm.
Assuming that the input signal is stationary.

Where is the optimal forward linear prediction coefficient vector

is the auto-correlation matrix of input signal

The Pseudo Affine Projection Relation is given by
is input signal, is desired output signal

In Fast Affine Projection algorithm the relation given by

can be written as





( ) ( ) 0,...0 , ... 1
1 1 , t t t
U X R
T
N N
a a =

| |
T
N
a a
1 1
.. 1

N , t
R
i
N
i
i
a

+ =
t t t
X X U
1
1
t
x
t
u
( ) ( ) 0,...0 , ... 1
1 1 , t t t
U X R
T
N N
a a =

b P
N t
=
,
R
Gauss-Seidel Pseudo Affine Projection algorithm (GSPAP)
Where the following relations is used

is 1-D vector of length with only one non-zero element at the top of the vector
which is taken as unity.

We solve this linear system using one Gauss-Seidel iteration procedure.

The solution thus obtained is given by

is the desired output signal vector

The filter coefficient update equation is given by








T
T
N
T T
a a
P
(

=

t t t t t t
U X U X U X
1 1
,..., ,
1
b
i
N
i
i
P
P

=
t t
X U
1
0 0
1
t
U
( )
1 t t
t
H X U
X U
H H


+
+ =
T
t t
T
t
t t
y
o

1

Double-talk Detection

When v(n) is active, w(n) may diverge
DTD: Detect double-talk quickly
Distinguish double-talk from echo path variations
The Proposed DTD Algorithm
1) Form a error vector with its initial values equal to zero.
2) Keep the new error sample calculated as the difference between the original echo and output of the
adaptive FIR filter of AEC, on the top of the error vector like what we do for the signal vector.
3) Find the norm of the error vector i.e. calculate the error energy or variance.
4) If the error norm is less than a minimum threshold value (say EPSILON = 0.000001), then make the
error norm equal to that value to not to let the norm of the error vector equal to zero or a too low value
5) If the norm of the error vector is less than 0.1 times the norm of the signal vector, which happens when
near end speaker is silent, then the effective step size is made equal to step size times inverse of the
norm of the signal vector as what is done in the NLMS based AEC.
6) If the norm of the error vector is more than 0.1 times the norm of the signal vector, which happens
when near end speaker is speaking, then the step size is made equal to the product of step size and
the inverse of the four times of the sum of the input vector and the error vector norms.
7) Filter coefficient update is done using the update step-size modified in accordance with step 5 or step 6
depending upon whether single talk or double talk scenario.
8) After filter update, the NLP is implemented if the double talk is not detected.
This algorithm is simple, robust and is very quick in determining the presence of double talk or not.
In the simulation, initially single talk situation exists and after a specified duration, near speech signal
which is smaller in length to the far end signal is added to the echo thus error signal contains a mixture
of residual echo and the near end speech signal. After the near end speech signal is over, again the far
end signal alone remains and the error signal containing only the residual echo is observed.
The output signal is recorded and is observed using the WAVOSAUR software. It can be observed
that there exists no divergence of the residual echo during the double talk period and also the
convergence time of the residual echo, after the near end speech signal ends, is also very less.




Fig5:The flowchart for the simulation of the echo canceller algorithm
start
Get far end signal, x(n)
Create echo signal, r(n) from
far end signal
Get near end signal, v(n)
Combine r(n) and v(n) to obtain
Desired signal d(n)
does double talk
exist
Filter loop, filter
Coefficients are frozen

Nlms loop update filter coefficients
Subtract estimated echo from d(n)
to produce residual error signal, e(n)
Do non linear
Processing to
Remove residual
echo
no
YES
Matlab simulation and results
Fig 6 -The far-end signal, echo signal and the residual echo

Fig 7: Far end signal, Residual Echo Signal, error signal for the NLMS method
based Acoustic Echo Cancellation System

Fig 8: Far end signal, Residual Echo Signal, error signal for the VSS-NLMS method based
Acoustic Echo Cancellation System
Fig 9 : Far end signal, Residual Echo Signal, error signal for the GNGD VSS-NLMS method
based Acoustic Echo Cancellation System

Fig 10: Far end signal, Residual Echo Signal, error signal for the GSER VSS-NLMS
method based Acoustic Echo Cancellation System
Comparison between NLMS and Pseudo Affine Projection based Acoustic
Echo Cancellation System
Fig 10: Residual Echo Signal for the NLMS based Acoustic Echo Cancellation System
Fig 11: Residual Echo Signal for the Pseudo Affine Projection method based Acoustic
Echo Cancellation System
Implementation
Software requirements - code compositor studio, visual
studio

System requirements
- IBM pc
- windows XP
- 32 Mbytes RAM and 100 Mbytes hard
disk space

Hardware requirements
-TMS320C6713dsk Digital Signal Processor
- USB cable

Code compositor studio

The Code Composer Studio integrated development environment is used to
build and debug embedded real-time software applications

CCS provides an IDE to incorporate the software tools. it includes tools for
code generation, such as a C compiler, an assembler, and a linker.

CCS includes the following components
- TMS320C67x code generation tools
- CCS IDE
- DSP/BIOS plug-ins and API
- RTDX plug-in, host interface, and API

Code Composer Studio Development process- Code Composer Studio extends
the basic code generation tools with a set of debugging and real-time analysis
capabilities. Code Composer Studio supports all phases of the development
cycle shown here:


Fig 12: code compositor studio setup
Code generation tools

Fig 13: Software development flow
1) Code generation tools
The C compiler
The assembler
The linker
The archiver
assembly translator utility
library-build utility
The run-time-support libraries
The hex conversion utility
The cross-reference lister
The absolute lister

2) CCS Integrated Development Environment (IDE)
The Code Composer Studio Integrated Development Environment (IDE) is
designed to allow you to edit, build, and debug DSP target programs.

3) DSP/BIOS is a scalable real-time kernel. It is designed for applications that require real-time
scheduling and synchronization, host-to-target communication, or real-time instrumentation.
4) RTDX plug-in, host interface, and API- Real-Time Data Exchange (RTDX) provides real-
time, continuous visibility into the way target applications operate in the real world. RTDX
allows system developers to transfer data between a host and target devices without interfering
with the target application


Programming in CCS
Developing a Simple Program on CCS
Debugging a Project using CCS
Additional features of CCS
Creating a New Project
Adding Files to a Project
Building and Running the Program
Using Break Points and the Watch Window
Adding a Probe Point for File I/O
Displaying Graphs

TMS320C6713dsk
Digital signal processors such as the TMS320C6x (C6x) family of processors are
like fast special-purpose microprocessors with a specialized type of architecture and
an instruction set appropriate for signal processing.

The C6x notation is used to designate a member of Texas Instruments (TI)
TMS320C6000 family of digital signal processors.

Based on VLIW architecture and floating point processor.

Typical Applications for the TMS320 Family in telecommunications, automotive,
control system, military, medical, instrumentation and in image processing.

Package Contents of tms320c6713dsk

Block diagram of TMS320C6713 DSK
Fig 15 : block diagram of TMS320C6713dsk
TMS320C6713 DSP features
A Texas Instruments TMS320C6713 DSP operating at 225 MHz

An AIC23 stereo codec

8 Mbytes of synchronous DRAM

512 Kbytes of non-volatile Flash memory (256 Kbytes usable in default
configuration)

4 user accessible LEDs and DIP switches

Software board configuration through registers implemented in CPLD

Configurable boot options

JTAG emulation through on-board JTAG emulator with USB host

Single voltage power supply (+5V)
Fig 16- Board Diagram of TMS320C6713 DSK

Board Components
CPLD (Programmable Logic)
AIC23 Codec
Synchronous DRAM
Flash Memory
LEDs and DIP Switches
Daughter Card Interface
Block diagram of hardware

ERLE curve for NLMS,VSS nlms based AEC with white
Gaussian noise samples as the far-end input
NLMS
VSS-NLMS
ERLE curve for NLMS,VSS nlms based AEC with white
Gaussian noise samples as the far-end input
RR VSS-NLMS
GNGD VSS-NLMS
ERLE curve for NLMS,VSS nlms based AEC with white
Gaussian noise samples as the far-end input
GESR VSS-NLMS
NSCE curve for NLMS,VSS-NLMS based AEC with white
Gaussian noise samples as the far-end input
NLMS
VSS-NLMS
NSCE curve for NLMS,VSS-NLMS based AEC with white
Gaussian noise samples as the far-end input
RR VSS-NLMS
GNGD VSS-NLMS
NSCE curve for NLMS,VSS-NLMS based AEC with white
Gaussian noise samples as the far-end input
GESR VSS-NLMS
Conclusion and future work
The Scope of this project is to implement Acoustic Echo Cancellers based on various
adaptive filtering methods.The implementations are to be done using
TMS320C6713DSK for each AEC and the performances have to be evaluated for
comparison.

Many parameters occur in algorithm such as step size, regularization parameter, safety
constant etc used to achieve target performance. the parameter tuning gains an
important role in simulation process.
In recent applications like acoustic echo cancellation, the order of the impulse response
to be estimated is very high, and these traditional approaches are inefficient and real
time implementation becomes difficult. .

This approach, referred to as sub-band adaptive filtering, is expected to reduce not only
the computational complexity but also to improve the convergence rate of the adaptive
algorithm

But in practice, different sub-band adaptive algorithms have to be used to enhance the
performance with respect to complexity, convergence rate and processing delay.

A single sub-band adaptive filtering algorithm which outperforms the full band scheme
in all applications is yet to be realized.

The Perfect Reconstruction Filter Bank (PRFB) is used to model the linear FIR system.
The structure offers efficient implementation with reduced arithmetic complexity.


references
1.Srinivasaprasath Raghavendran Implementation of an Acoustic Echo Canceller Using Matlab
2. ITU-Ts G.167 Acoustic Echo Cancellers
3. Lu Lu Implementation of Acoustic Echo Cancellation for PC Applications
4. Ahmed I. Sulyman & Azzedine Zerguine Echo Cancellation Using a Variable Step-Size
NLMS Algorithm
5. Junghsi Lee, Jia-Wei Chen, and Hsu-Chang Huang Performance Comparison of
Variable Step-Size NLMS Algorithms
6. Y. S. Choi, H. C. Shin, and W. J. Song, Robust regularization for normalized LMAlgorithms,
IEEETransactions on Circuits and Systems II, Express Briefs, Vol. 53, No. 8, pp. 627
631, Aug. 2006.
7. D. P. Mandic, A generalized normalized gradient descent algorithm, IEEE Signal Processing
Letters, Vol. 11, No. 2, pp. 115118, Feb. 2004.
8. J. Lee, H. C. Huang, and Y. N. Yang, The generalized square-error-regularized LMS
Algorithm, Proceedings of WCECS 2008, pp. 157 160, Oct. 2008.
9. F. Bouteille, P. Scalart, M. Corazza, Pseudo Affine Projection Algorithm New Solution for
Adaptive Identification, Eurospeech, 1999
10. Texas Instruments, TMS320C6713 DSK, Floating-Point Digital Signal Processors, Data
Sheet, Dallas, TX, June 2006
11. Spectrum Digital, TMS320C6713 DSK Technical Reference, INC. 12502 Exchange Drive,
Suite, November 2003
12. TIs Code Composer Studio (CCS) Users Guide